Systems and methods for processing an audio signal for replay on an audio device

ABSTRACT

Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using band pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path. Finally, the feedforward-compressed subband signals are recombined.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation-in-part of U.S. application Ser. No.16/244,727 filed Jan. 10, 2019 and entitled “SYSTEMS AND METHODS FORPROCESSING AN AUDIO SIGNAL FOR REPLAY ON AN AUDIO DEVICE”, which claimspriority to U.S. Pat. No. 10,199,047 issued on Feb. 5, 2019 and thedisclosures of which are both herein incorporated by reference in theirentirety.

FIELD OF INVENTION

This invention relates generally to the field of digital signalprocessing (DSP), audio engineering and audiology—more specificallysystems and methods for processing an audio signal for replay on anaudio device, for example for providing an enhanced listening experienceon the audio device.

BACKGROUND

Traditional DSP sound personalization methods often rely on equalizationtechniques that apply compensatory frequency gain according to a user'shearing profile (see e.g. U.S. Pat. Nos. 9,138,178, 9,468,401 9,680,438,9,898,248). Typically, a pure tone threshold (PTT) hearing test isemployed to identify frequencies in which a user exhibits raised hearingthresholds. Based on the audiogram data, the frequency output is thenmodulated accordingly. In this regard, the approach to augmenting thesound experience for the user is one dimensional. The gain may enablethe user to recapture previously unheard frequencies, however they maysubsequently experience loudness discomfort. Listeners withsensorineural hearing loss typically have similar, or even reduced,discomfort thresholds when compared to normal hearing listeners, despitetheir hearing thresholds being raised. To this extent, their dynamicrange is narrower and simply adding EQ gain would be detrimental totheir hearing health in the long run (FIG.1).

Dynamic range compression (DRC) can be used to address this issue byamplifying quieter sounds while reducing the volume of loud sounds, thusnarrowing the dynamic range of the audio. However, this could pose aproblem, as a low frequency rumble could prevent amplification of a highfrequency sound of interest. For this reason, hearing aid processorsemploy multichannel DRC where the faintest sounds are amplifiedconsiderably, but where high-intensity sounds are not. To this extent,conventional hearing aids are designed for use in real world situationswhere a wide dynamic range of sounds are relevant to the listener, i.e.the listener wants to make sense of sonic information such as aloud-voiced person speaking in front of them, while at the same timethey want to be able to detect the faint sound of a car approaching themfrom a distance while walking down the street. Although this works forpractical, real world matters, audio content consumed on mobile devices,or other similar devices, have very different signal statistics to thesounds that someone will encounter in their daily life, so a differentprocessing strategy is required to provide the listener with abeneficial sound personalization experience.

The ability to digitally recreate the functional processing of healthyhuman hearing would enable a more natural and clear listening experiencefor a hearing impaired (HI) user. Only until recently has the physics ofthe human ear been better understood. The human ear pre-processes soundsinto a format that is optimal for transmission to the brain to makesense of the sonic environment. The pre-processing can be modelled as anumber of hierarchical signal processes and feedback loops, many ofwhich are non-linear, resulting in a complex, non-linear system.Although hearing loss typically begins at higher frequencies, listenerswho are aware that they have hearing loss do not typically complainabout the absence of high frequency sounds. Instead, they reportdifficulties listening in a noisy environment and in hearing out thedetails in a complex mixture of sounds, such as in orchestral music. Inessence, off frequency sounds more readily mask information with energyin other frequencies for HI individuals—music that was once clear andrich in detail becomes muddled. This is because music itself is highlyself-masking.

As hearing deteriorates, the signal-conditioning capabilities of the earbegin to break down, and thus HI listeners need to expend more mentaleffort to make sense of sounds of interest in complex acoustic scenes(or miss the information entirely). A raised threshold in an audiogramis not merely a reduction in aural sensitivity, but a result of themalfunction of some deeper processes within the auditory system thathave implications beyond the detection of faint sounds.

Recent studies attempted to better model the physics of the human ear,modelling the interconnection of the basilar membrane, the medialolivocochlear complex and the inner and outer hair cells within themiddle ear. Building on hearing aid format technology, Clark et al.(2012) developed an algorithm to better model human hearing, mimickingthe attenuation effect of the medial olivocochlear to the basilarmembrane, which data from the aforementioned suggests might improvespeech-in-noise robustness (see: Clark et al., A frequency-selectivefeedback model of auditory efferent suppression and its implications forthe recognition of speech in noise. Journal of the acoustical society ofAmerica, Volume 132, issue 3, pages 1535 to 1541, 2012). This result isachieved by implementing a delayed feedback attenuation control (DFAC)to a dual resonance non-linear (DRNL) algorithm within a spectrallydecomposed system (for DRNL see: E. Lopez-Poveda and R. Meddis. A humannonlinear cochlear filterbank. Journal of the acoustical society ofAmerica, Volume 110, issue 6, Pages 3107 to 3118, 2001). The DRNLalgorithm includes instantaneous dynamic range compression.

However, this algorithm served merely as a framework for modeling thehearing system and was not specifically designed for sound augmentation.To this extent, it has some drawbacks on the subjective hearingexperience caused by the lack of control over the distortion products.These include a reduced ability to control distortion, a limitedfrequency resolution and phase distortion that can cause temporalsmearing of sound (if used in combination with narrowband filters) andtherefore reduced fidelity of the reconstructed output, resulting in aless clear listening experience for the user. Namely, although thisalgorithm could potentially improve some aspects of real-world use casesif used by hard of hearing users, it would fail to improve the listeningexperience for a broader category of listeners in the context of audio.Accordingly, it is the object of this invention to create an improved,biologically-inspired DSP that provides a listener with beneficial soundpersonalization.

SUMMARY OF THE INVENTION

The problems raised in the known prior art will be at least partiallysolved in the invention as described below. The features according tothe invention are specified within the independent claims, advantageousimplementations of which will be shown in the dependent claims. Thefeatures of the claims can be combined in any technically meaningfulway, and the explanations from the following specification as well asfeatures from the figures which show additional embodiments of theinvention can be considered.

By creating improved, biologically-inspired DSP algorithms that moreclosely mimic the functional processing of the healthy human ear, thepresented technology solves the limitations inherent in prior art DSPmethodologies, namely poor frequency resolution and temporal smearingcaused by group delay differences between bands. To this extent, theinvention provides an enhanced listening experience on an audio devicefor both hard of hearing listeners as well as individuals with low tomoderate hearing loss, who experience clearer listening experience.

In general, the technology features methods for processing an audiosignal for replay on an audio device. In particular, the methods may bemethods of processing an audio signal to provide an enhanced hearingexperience on (e.g., when replayed on) an audio device.

According to an aspect, a method of processing an audio signal forreplay on an audio device may include a) spectral decomposition of theaudio signal into a plurality of frequency bands (e.g., into a pluralityof subband signals, each subband signal in a respective frequency band)using a bandpass filter (e.g., an input bandpass filter). The method mayfurther include b) for each frequency band, providing the audio signalin the frequency band (e.g., the subband signal) to a respectivemodulator and from the modulator output, providing the audio signal inthe frequency band to a respective first dynamic range compressor (e.g.,to a feedforward DRC as an example of the first DRC). The feedforwardDRC may be part of a first processing path for the respective frequencyband (or subband signal), and the (modulated) audio signal in thefrequency band (e.g., the subband signal) may be provided to the firstprocessing path. The first processing path may be referred to as afeedforward path. The method may further include c) for each frequencyband, feedforward compressing the modulated audio signal in thefrequency band (e.g., the modulated subband signal) to obtain afeedforward-compressed audio signal in the respective frequency band(e.g., a feedforward-compressed subband signal). Therein, thefeedforward DRC is slowed relative to an instantaneous DRC. This slowingmay be done either directly or indirectly. By virtue of this slowing ofdynamic range compression, the spectral spread of harmonic distortionand intermodulation distortion products can be controlled. Thefeedforward compressed frequency band (e.g., the feedforward-compressedaudio signal in the frequency band, or the feedforward-compressedsubband signal) may be provided to a respective compression output. Themethod may further include) for each frequency band, feedbackcompressing each feedforward compressed frequency band from therespective compression output. To this end, each feedforward compressedfrequency band may be provided to a respective second processing paththat includes a respective second DRC. The second processing path may bereferred to as a feedback path. The second DRC may be referred to as afeedback DRC (as an example of the second DRC). Further, the feedbackDRC may be delayed relative to the feedforward DRC. That is, the outputof the second processing path may be deliberately delayed, e.g., by adelay element (such a s a buffer, for example). The delay may beinserted before or after the feedback DRC. The (delayed) feedbackcompressed frequency band is then provided to the modulator for therespective frequency band. In general, the output of the secondprocessing path is provided to the respective modulator. The modulatormay operate in dependence on (e.g., under control of) the output of thesecond processing path. The modulator may provide attenuation at thecompression input of step c, in dependence on the output of the secondprocessing path. The method may further include e) recombining thefeedforward-compressed frequency bands (e.g., the feedforward-compressedaudio signals in the frequency bands, or the feedforward-compressedsubband signals).

Configured as above, the proposed method has the advantage and technicaleffect of providing an enhanced listening experience for a user. This isachieved by processing an audio signal using techniques that mimic thefunctional processing of the healthy human auditory system.

In one embodiment, the input band pass filter is phase linear. In afurther embodiment, the phase linear input band pass filter is a finiteimpulse response filter operating in the frequency domain.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to oversampling the respective subband signal. Forexample, the oversampling may comprise applying an n-point FFT to thesubband signal and overlapping the FFT transforms by n/N samples, whereN is the oversampling rate and n is larger than N. For a given n, theoversampling rate N may range from 2 to n/2, for example (whichtranslates into an overlap in the range between n/2 samples and 2samples). For typical implementations (e.g., n=256, 512, 1024), theoversampling rate N may be in the range from 128 to 512, for example.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to increasing attack and/or release time constants of thefirst DRC (i.e., setting the attack and/or release time constants tovalues different from 0). For example, slowing the first DRC relative toan instantaneous DRC may relate to setting attack and/or release timeconstants of the first DRC based on a time constant τ that is selectedfrom a range extending from 0.01 ms to 3 ms. Likewise, slowing the firstDRC relative to an instantaneous DRC may relate to setting attack and/orrelease time constants of the first DRC based on a time constant τ thatcorresponds to a frequency f within the respective (frequency) subband.This frequency may be the lower cutoff frequency, the upper cutofffrequency, or the center frequency of the subband. In some embodiments,the aforementioned attack and/or release time constants could beprovided as independent values t_(A) and t_(B), for example, that arenot based on a time constant τ.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to both oversampling the respective subband signal aswell as to increasing attack and/or release time constants of the firstDRC (i.e., setting the attack and/or release time constants to valuesdifferent from 0). One of ordinary skill in the art may appreciate thatslowing the first DRC may be achieved through the combination of bothindirect slowing (i.e. oversampling) and direct slowing (i.e. alteringthe time constants of the first DRC). For direct slowing, τ is relatedto the cutoff frequency f_(c), an alternative parameter of the RCcircuit, by τ=RC=1/(2π f_(c)). An (indirect) equivalent of the timeconstant τ of the directly slowed first DRC for the slowing byoversampling can be calculated by dividing the oversampling rate N bythe sampling rate (e.g. 44100 Hz). To this extent, the combinatorialeffect of indirect and direct slowing of the DRC is readily calculableas a function of these two values.

According to another aspect, a method of processing an audio signal toprovide an enhanced hearing experience on (e.g., when replayed on) anaudio device may comprise dividing an unprocessed audio signal into afirst signal pathway and a second signal pathway, processing the audiosignal in the first signal pathway, and recombining outputs of the firstsignal and second signal pathways at a ratio. The processing in thefirst signal pathway may include a) performing a spectral decompositionof the audio signal in the first signal pathway into a plurality ofsubband signals using a band pass filter. The processing in the firstsignal pathway may further include b) for each subband signal, providingthe subband signal to a respective modulator and from the modulatoroutput, providing the subband signal to a respective first processingpath that includes a first dynamic range compressor, DRC. The firstprocessing path may be referred to as a feedforward path. The processingin the first signal pathway may further include c) for each subbandsignal, feedforward compressing the subband signal by the respectivefirst DRC to obtain a feedforward-compressed audio signal in therespective frequency band (e.g., a feedforward compressed subbandsignal). Therein, the feedforward DRC is slowed relative to aninstantaneous DRC. This slowing may be done either directly orindirectly. By virtue of this slowing of dynamic range compression, thespectral spread of harmonic distortion and intermodulation distortionproducts can be controlled. The feedforward compressed frequency band(e.g., the feedforward-compressed audio signal in the frequency band, orthe feedforward-compressed subband signal) may be provided to arespective compression output. The processing in the first signalpathway may further include) feedback compressing each feedforwardcompressed frequency band from the respective compression output. Tothis end, each feedforward compressed frequency band may be provided toa respective second processing path that includes a respective secondDRC. The second processing path may be referred to as a feedback path.The second DRC may be referred to as a feedback DRC (as an example ofthe second DRC). Further, the feedback DRC may be delayed relative tothe feedforward DRC. That is, the output of the second processing pathmay be deliberately delayed, e.g., by a delay element (such a s abuffer, for example). The delay may be inserted before or after thefeedback DRC. The (delayed) feedback compressed frequency band is thenprovided to the modulator for the respective frequency band. In general,the output of the second processing path is provided to the respectivemodulator. The modulator may operate in dependence on (e.g., undercontrol of) the output of the second processing path. The modulator mayprovide attenuation at the compression input of step c, in dependence onthe output of the second processing path. The processing in the firstsignal pathway may further include e) recombining thefeedforward-compressed frequency bands (e.g., the feedforward-compressedaudio signals in the frequency bands, or the feedforward-compressedsubband signals).

In one embodiment, the input band pass filter is phase linear. In afurther embodiment, the phase linear input band pass filter is a finiteimpulse response filter operating in the frequency domain.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to oversampling the respective subband signal. Forexample, the oversampling may comprise applying an n-point FFT to thesubband signal and overlapping the FFT transforms by n/N samples, whereN is the oversampling rate and n is larger than N. For a given n, theoversampling rate N may range from 2 to n/2, for example (whichtranslates into an overlap in the range between n/2 samples and 2samples). For typical implementations (e.g., n=256, 512, 1024), theoversampling rate N may be in the range from 128 to 512, for example.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to increasing attack and/or release time constants of thefirst DRC (i.e., setting the attack and/or release time constants tovalues different from 0). For example, slowing the first DRC relative toan instantaneous DRC may relate to setting attack and/or release timeconstants of the first DRC based on a time constant τ that is selectedfrom a range extending from 0.01 ms to 3 ms. Likewise, slowing the firstDRC relative to an instantaneous DRC may relate to setting attack and/orrelease time constants of the first DRC based on a time constant τ thatcorresponds to a frequency f within the respective (frequency) subband.This frequency may be the lower cutoff frequency, the upper cutofffrequency, or the center frequency of the subband. In some embodiments,the aforementioned attack and/or release time constants could beprovided as independent values t_(A) and t_(B), for example, that arenot based on a time constant τ.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to both oversampling the respective subband signal aswell as to increasing attack and/or release time constants of the firstDRC (i.e., setting the attack and/or release time constants to valuesdifferent from 0). One of ordinary skill in the art may appreciate thatslowing the first DRC may be achieved through the combination of bothindirect slowing (i.e. oversampling) and direct slowing (i.e. alteringthe time constants of the first DRC). For direct slowing, τ is relatedto the cutoff frequency f_(c), an alternative parameter of the RCcircuit, by τ=RC=1/(2π f_(c)). An (indirect) equivalent of the timeconstant τ of the directly slowed first DRC for the slowing byoversampling can be calculated by dividing the oversampling rate N bythe sampling rate (e.g. 44100 Hz). To this extent, the combinatorialeffect of indirect and direct slowing of the DRC is readily calculableas a function of these two values.

According to another aspect, in a method of processing an audio signalto provide an enhanced hearing experience on (e.g., when replayed on) anaudio device, the subband signal itself may further be split into afirst signal pathway and a second signal pathway and recombined at aratio. Thus, the method may include a) performing a spectraldecomposition of the audio signal into a plurality of subband signalsusing a band pass filter. The method may further include b) for eachsubband signal, dividing the subband signal into a first signal pathwayand a second signal pathway, processing the subband signal in the firstsignal pathway, and recombining the first and second signal pathways ata ratio to obtain a processed subband signal. Processing the subbandsignal in the first signal pathway may comprise b1) providing the audiosignal in the frequency band (e.g., the subband signal) to a respectivemodulator and from the modulator output, providing the audio signal inthe frequency band to a respective first dynamic range compressor (e.g.,to a feedforward DRC as an example of the first DRC). The feedforwardDRC may be part of a first processing path for the respective frequencyband (or subband signal), and the (modulated) audio signal in thefrequency band (e.g., the subband signal) may be provided to the firstprocessing path. The first processing path may be referred to as afeedforward path. Processing the subband signal in the first signalpathway may further comprise b2) feedforward compressing the audiosignal in the frequency band (e.g., the modulated subband signal) by therespective first DRC to obtain a feedforward-compressed audio signal(e.g., a feedforward-compressed subband signal). Therein, thefeedforward DRC is slowed relative to an instantaneous DRC. This slowingmay be done either directly or indirectly. By virtue of this slowing ofdynamic range compression, the spectral spread of harmonic distortionand intermodulation distortion products can be controlled. Thefeedforward compressed frequency band (e.g., the feedforward-compressedaudio signal in the frequency band, or the feedforward-compressedsubband signal) may be provided to a respective compression output.Processing the subband signal in the first signal pathway may furthercomprise b3) feedback compressing the feedforward compressed frequencyband from the respective compression output. To this end, thefeedforward compressed frequency band may be provided to a respectivesecond processing path that includes a respective second DRC. The secondprocessing path may be referred to as a feedback path. The second DRCmay be referred to as a feedback DRC (as an example of the second DRC).Further, the feedback DRC may be delayed relative to the feedforwardDRC. That is, the output of the second processing path may bedeliberately delayed, e.g., by a delay element (such a s a buffer, forexample). The delay may be inserted before or after the feedback DRC.The (delayed) feedback compressed frequency band is then provided to themodulator for the respective frequency band. In general, the output ofthe second processing path is provided to the respective modulator. Themodulator may operate in dependence on (e.g., under control of) theoutput of the second processing path. The modulator may provideattenuation at the compression input of step b3, in dependence on theoutput of the second processing path. The method may further include c)recombining the feedforward-compressed frequency bands (e.g., thefeedforward-compressed audio signals in the frequency bands, or thefeedforward-compressed subband signals).

In one embodiment, the input band pass filter is phase linear. In afurther embodiment, the phase linear input band pass filter is a finiteimpulse response filter operating in the frequency domain.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to oversampling the respective subband signal. Forexample, the oversampling may comprise applying an n-point FFT to thesubband signal and overlapping the FFT transforms by n/N samples, whereN is the oversampling rate and n is larger than N. For a given n, theoversampling rate N may range from 2 to n/2, for example (whichtranslates into an overlap in the range between n/2 samples and 2samples). For typical implementations (e.g., n=256, 512, 1024), theoversampling rate N may be in the range from 128 to 512, for example.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to increasing attack and/or release time constants of thefirst DRC (i.e., setting the attack and/or release time constants tovalues different from 0). For example, slowing the first DRC relative toan instantaneous DRC may relate to setting attack and/or release timeconstants of the first DRC based on a time constant τ that is selectedfrom a range extending from 0.01 ms to 3 ms. Likewise, slowing the firstDRC relative to an instantaneous DRC may relate to setting attack and/orrelease time constants of the first DRC based on a time constant τ thatcorresponds to a frequency f within the respective (frequency) subband.This frequency may be the lower cutoff frequency, the upper cutofffrequency, or the center frequency of the subband.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to both oversampling the respective subband signal aswell as to increasing attack and/or release time constants of the firstDRC (i.e., setting the attack and/or release time constants to valuesdifferent from 0). One of ordinary skill in the art may appreciate thatslowing the first DRC may be achieved through the combination of bothindirect slowing (i.e. oversampling) and direct slowing (i.e. alteringthe time constants of the first DRC). For direct slowing, τ is relatedto the cutoff frequency f_(c), an alternative parameter of the RCcircuit, by τ=RC=1/(2π f_(c)). An (indirect) equivalent of the timeconstant τ of the directly slowed first DRC for the slowing byoversampling can be calculated by dividing the oversampling rate N bythe sampling rate (e.g. 44100 Hz). To this extent, the combinatorialeffect of indirect and direct slowing of the DRC is readily calculableas a function of these two values.

According to another aspect, a method of processing an audio signal toprovide an enhanced hearing experience on (e.g., when replayed on) anaudio device may include a) spectral decomposition of an audio signalinto a plurality of frequency bands (e.g., into a plurality of subbandsignals, each subband signal in a respective frequency band) using abandpass filter (e.g., an input bandpass filter). The method may furtherinclude b) for each frequency band, providing the audio signal in thefrequency band (e.g., the subband signal) to a respective modulator andfrom the modulator output, providing the audio signal in the frequencyband to a respective first dynamic range compressor (e.g., to afeedforward DRC as an example of the first DRC). The feedforward DRC maybe part of a first processing path for the respective frequency band (orsubband signal), and the (modulated) audio signal in the frequency band(e.g., the subband signal) may be provided to the first processing path.The first processing path may be referred to as a feedforward path. Themethod may further include c) for each frequency band, feedforwardcompressing the modulated audio signal in the frequency band (e.g., themodulated subband signal) to obtain a feedforward-compressed audiosignal in the respective frequency band (e.g., a feedforward-compressedsubband signal). Therein, the feedforward DRC is slowed relative to aninstantaneous DRC. This slowing may be done either directly orindirectly. By virtue of this slowing of dynamic range compression, thespectral spread of harmonic distortion and intermodulation distortionproducts can be controlled. The feedforward compressed frequency band(e.g., the feedforward-compressed audio signal in the frequency band, orthe feedforward-compressed subband signal) may be provided to arespective compression output. The method may further include) for eachfrequency band, providing the feedforward-compressed subband signal to asecond processing path that includes a second DRC and further providingone or more feedforward-compressed subband signals from neighboringfrequency bands, each weighted with a respective weighting factor, tothe second processing path. Therein, in the second processing path, thefeedforward-compressed subband signal and the weightedfeedforward-compressed subband signals from the neighboring frequencysubbands are compressed by the respective second DRC. The second DRC maybe referred to as a feedback DRC (as an example of the second DRC).Further, the feedback DRC may be delayed relative to the feedforwardDRC. That is, the output of the second processing path may bedeliberately delayed, e.g., by a delay element (such a s a buffer, forexample). The delay may be inserted before or after the feedback DRC.The (delayed) feedback compressed frequency band is then provided to themodulator for the respective frequency band. In general, the output ofthe second processing path is provided to the respective modulator. Themodulator may operate in dependence on (e.g., under control of) theoutput of the second processing path. The modulator may provideattenuation at the compression input of step c, in dependence on theoutput of the second processing path. The method may further include e)recombining the feedforward-compressed frequency bands (e.g., thefeedforward-compressed audio signals in the frequency bands, or thefeedforward-compressed subband signals).

In one embodiment, the input band pass filter is phase linear. In afurther embodiment, the phase linear input band pass filter is a finiteimpulse response filter operating in the frequency domain.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to oversampling the respective subband signal. Forexample, the oversampling may comprise applying an n-point FFT to thesubband signal and overlapping the FFT transforms by n/N samples, whereN is the oversampling rate and n is larger than N. For a given n, theoversampling rate N may range from 2 to n/2, for example (whichtranslates into an overlap in the range between n/2 samples and 2samples). For typical implementations (e.g., n=256, 512, 1024), theoversampling rate N may be in the range from 128 to 512, for example.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to increasing attack and/or release time constants of thefirst DRC (i.e., setting the attack and/or release time constants tovalues different from 0). For example, slowing the first DRC relative toan instantaneous DRC may relate to setting attack and/or release timeconstants of the first DRC based on a time constant τ that is selectedfrom a range extending from 0.01 ms to 3 ms. Likewise, slowing the firstDRC relative to an instantaneous DRC may relate to setting attack and/orrelease time constants of the first DRC based on a time constant τ thatcorresponds to a frequency f within the respective (frequency) subband.This frequency may be the lower cutoff frequency, the upper cutofffrequency, or the center frequency of the subband.

In some embodiments, slowing the first DRC relative to an instantaneousDRC may relate to both oversampling the respective subband signal aswell as to increasing attack and/or release time constants of the firstDRC (i.e., setting the attack and/or release time constants to valuesdifferent from 0). One of ordinary skill in the art may appreciate thatslowing the first DRC may be achieved through the combination of bothindirect slowing (i.e. oversampling) and direct slowing (i.e. alteringthe time constants of the first DRC). For direct slowing, τ is relatedto the cutoff frequency f_(c), an alternative parameter of the RCcircuit, by τ=RC=1/(2π f_(c)). An (indirect) equivalent of the timeconstant τ of the directly slowed first DRC for the slowing byoversampling can be calculated by dividing the oversampling rate N bythe sampling rate (e.g. 44100 Hz). To this extent, the combinatorialeffect of indirect and direct slowing of the DRC is readily calculableas a function of these two values.

DSP methods that mimic the functional processing of the healthy humanare vital to the creation of objectively and subjectively enrichedlistening experiences. By more precisely recreating the actions of thebasilar membrane and MOC through sound processing, both the HI andhealthy listener enjoy a clearer, more natural sound on an audio device.

Unless otherwise defined, all technical terms used herein have the samemeaning as commonly understood by one of ordinary skill in the art towhich this technology belongs.

The term “audio device”, as used herein, is defined as any device thatoutputs audio, including, but not limited to: mobile phones, computers,televisions, hearing aids, headphones and/or speaker systems.

The phrase “dynamic range compression” or “DRC”, as used herein, isdefined as an audio process that reduces the dynamic range of an audiosignal. A compressor may either have a feedforward or feedback design.Dynamic range compression may occur instantaneously or the rate may becontrolled through adjustment of the compressor's attack and releasetime constants.

The phrase “indirectly slowed”, as used herein, is defined as theslowing of a DRC through means other than adjusting a compressor'sattack and release times, such as through multi-rate signal processing.

The phrase “directly slowed”, as used herein, is defined as the slowingof a DRC through adjusting a compressor's attack and release timeconstants.

The phrase “bandpass filter”, as used herein, is defined as a devicethat (substantially) passes frequencies within a certain range andattenuates frequencies outside that range.

The phrase “phase linear”, as used herein, is defined as propertywherein the phase response of a filter is a linear function of frequencysuch that all frequency components of an input signal are shifted intime by the same constant amount, resulting in no phase distortion.

The phrase “harmonic distortion”, as used herein, is defined as thegeneration of multiples of the original frequencies caused by anonlinear system.

The phrase “intermodulation distortion”, as used herein, is defined asthe generation of cross-product frequencies produced when two or moresignals mix in a nonlinear system.

The phrase “computer readable storage medium”, as used herein, isdefined as a solid, non-transitory storage medium including, but notlimited to: USB storages with flash memory, a CD-ROM, DVD or BluRay®, adisk or a tape. It may also be a physical storage place in a serveraccessible by a user, e.g. to download for installation of the computerprogram on her device or for cloud computing.

BRIEF DESCRIPTION OF THE DRAWINGS

In order to describe the manner in which the above-recited and otheradvantages and features of the disclosure can be obtained, a moreparticular description of the principles briefly described above will berendered by reference to specific embodiments thereof, which areillustrated in the appended drawings. Understand that these drawingsdepict only example embodiments of the disclosure and are not thereforeto be considered to be limiting of its scope, the principles herein aredescribed and explained with additional specificity and detail throughthe use of the accompanying drawings in which:

FIG. 1 illustrates a graph showing dynamic range reduction in hearingimpaired listeners;

FIG. 2 illustrates a processing model of the auditory system and theafferent and efferent pathways that affect hearing modulation;

FIG. 3 illustrates a prior art hearing aid circuit;

FIG. 4A illustrates the core digital signal processing circuit inspiredby the basilar membrane and the medial olivocochlear complex;

FIG. 4B illustrates a core digital signal processing circuit with adynamic threshold controller;

FIG. 4C illustrates a graph of various dynamic threshold DRCadjustments;

FIG. 4D illustrates a soft knee function that can be provided with afeedforward compressor;

FIG. 5A illustrates an example embodiment in which the signal isspectrally decomposed, parallelly compressed and recombined;

FIG. 5B illustrates an example embodiment in which an asymmetric softclipping function is applied to an output of the multiband compressorsystem;

FIG. 6 illustrates a spectrogram in which frequency distortion isreduced using the configuration of FIG. 5;

FIG. 7 illustrates an example of the biologically-designed DSPtechnology disclosed herein sharpening the psychometric tuning curve ofan HI subject;

FIG. 8 illustrates distortion patterns from two compressed test signals;when the test signals are shifted closer to the hypothetical edge of aband pass filter, the distortion pattern is more symmetrical when anoutput band pass filter is not employed;

FIG. 9 illustrates a further embodiment in which the signal is dividedinto a first signal pathway and a second signal pathway (a processed andan unprocessed pathway), and subsequently recombined at a user definedratio;

FIG. 10 illustrates a further embodiment in which the signal is dividedinto a first signal pathway and a second signal pathway, andsubsequently recombined at a user defined ratio;

FIG. 11 illustrates an additional embodiment, in which the signal isdivided into a first signal pathway and a second signal pathway in eachfrequency band, which can be subsequently recombined at various userdefined ratios according to the frequency band;

FIG. 12. illustrates a broader example of using the technique of usingthe structure of FIG. 11 in a spectrally decomposed audio signal;

FIG. 13. Illustrates an example wherein the control signal in thefeedback DRC may include weighted versions of the same signal inneighboring frequency bands;

FIG. 14 shows an example of a system for implementing certain aspects ofthe present technology.

DETAILED DESCRIPTION

Various example embodiments of the disclosure are discussed in detailbelow. While specific implementations are discussed, theseimplementations are for illustration purposes only. One of ordinaryskill in the art will recognize that other components and configurationsmay be used without parting from the spirit and scope of the disclosure.

In order to create audio processing algorithms that mimic the functionalprocessing of the human ear, a framework for healthy hearing must firstbe developed. Generally, the model of normal hearing consists ofcascading stages, simulating the physiological parts of the signalprocessing pathway in the auditory system. The model developed byMeddis, R. (see Meddis, R., N. R. Clark, W. Lecluyse, and T. Jürgens,“BioAid—A Biologically Inspired Hearing Aid”. Audiological Acoustics 52:2013, 148-152, 2013) aimed to provide a faithful representation ofauditory nerve firing patterns, as seen in the model of the auditoryperiphery in FIG. 2. Initially, the acoustic input signal isfiltered—mimicking the outer- and middle-ear 201 frequency-dependenttransfer of sound pressure to displacement of the stapes. Subsequently,the signal is decomposed into frequency bands using filter bank 202. Thefilter bank consists of a nonlinear path, modeling the contribution ofouter hair cells and a linear path, modeling the passive response of thebasilar membrane. Subsequent stages simulate stereo displacement, innerhair cell 203 potential fluctuations, and finally transmitter releaseinto the synaptic cleft 204 located between inner hair cells and theauditory nerve 205.

To model the ipsilateral acoustic reflex (AR) 206 and the ipsilateralmedial olivocochlear reflex (MOC) 207, two efferent feedback loops areadded. The MOC 207 feedback loop is tonotopically implemented as anattenuation of activity in the nonlinear path of the of the filterbank—the amount of attenuation is controlled by the total spikingactivity in the corresponding frequency band on the brainstem 208 level.The MOC attenuation of BM response builds up during steady portions ofan acoustic stimulus and decays with a time constant 50 ms following theoffset of the stimulus. A delay of 10 ms between the onset of a stimulusand the beginning MOC attenuation is used to mimic synaptic latencies.The acoustic reflex 206 is implemented as an attenuation of the stapesresponse on the total activity of all neurons.

This healthy hearing model formed the basis of Meddis et al.'s“BioAid-algorithm”, as illustrated in FIG. 3. At first, the audio signalis provided at a control input 301, which is then spectrally decomposed302 into a plurality of frequency bands by the input band pass filter303 (which is an infinite impulse response (IIR) filter that isrecursive, and, thus, introduces phase distortion). Each frequency bandis processed in the schematically shown plurality of parallel channelshaving all the same operators, though different parameters to it. Eachrespective frequency band is provided at a compression input 304, whichis feed forward compressed by an instantaneous dynamic range compressor305 (with time constant zero). From the compression output 306, theaudio signal is processed by a feedback DRC 307, wherein the feedbackDRC 307 is delayed relative to an instantaneous DRC 305. Subsequently,the compressed audio signal is modulated via a modulator 308 toattenuate the audio signal provided to the feedforward DRC.

The delayed feedback DRC processing is characterized by two adjustableparameters: a threshold parameter and a strength parameter. Thethreshold parameter specifies the level of the output from thefeedforward DRC, at which the feedback processing starts to work. Thestrength parameter, which governs the amount of attenuation applied whenthe feedback processing is active, is a scalar that is multiplied by theratio of the input to the feedback-processing process relative to thefeedback processing threshold in dB (thus giving the attenuation valuesin dB).

This compressed audio signal from the compression output 306 is passedthrough an output IIR band pass filter 309 to control the spectralspread of distortion. This secondary IIR filter 309 has the samepass-band as the input IIR band pass filter 303. Subsequently, thecompressed frequency bands are modulated by a gain 310 and, finally,recombined in an operator 311 to form a full wide band audio signalagain to be provided at the control output.

As discussed previously, this algorithm has drawbacks on the subjectivehearing experience caused by the use and arrangement of IIR filters 303,309 and the instantaneous DRC 305. This configuration leads to thespread of audible distortion, leading to a negative impact on perceivedquality, particularly for users with milder forms of hearing loss (seee.g. FIGS. 6,8). The algorithm was designed as a hearing aid algorithmto aid hard of hearing listeners in real world use cases (e.g. speech innoise), and thus would be a poor model for personalizing audio content.To this extent, it is the object of the invention to improve theobjective and subjective listening experiences for a broad array of NHand HI listeners.

In FIGS. 4A-B, example embodiments of the invention are illustrated. Inboth FIGS. 4A and 4B, a wide band audio signal is provided at processinginput 401 and then spectrally decomposed into a plurality of frequencybands (e.g., into a plurality of subband signals, each subband signal ina respective frequency band). Spectral decomposition may be performed bythe input band pass filter 402, although it is appreciated that one ormore additional input band pass filters can be utilized withoutdeparting from the scope of the present disclosure. For example, FIGS.5A-B depict spectral decomposition 501 as performed by a plurality ofbandpass filters 502, as will subsequently be discussed in greaterdepth. More generally, each frequency band of the spectral decompositionmay have a respective center frequency f₀, as well as a lower cutofffrequency and/or an upper cutoff frequency.

Returning to the discussion of FIG. 4A, each respective frequency bandis provided at a compression input 403, which is feed forward compressedby feedforward DRC 404. Feedforward DRC 404 (or the dynamic rangecompression that is applied by the DRC 404) is slowed relative to aninstantaneous DRC. This slowing may be direct or indirect. Indirectlyslowing a feedforward compression can be achieved through means otherthan adjusting a DRC's attack and/or release time constants. This mayoccur through the use of a finite impulse response filter followed bysignal oversampling, which has the net effect of slowing downfeedforward compression relative to an instantaneous DRC. In this case,the compressed frequency bands are downsampled, e.g. to the originalsample rate and/or sufficient rate for further processing and soundquality. This technique not only improves algorithmic computationalefficiency, but further provides the opportunity to parameterize thespread of distortion independently from the analysis bandwidth. Inanother embodiment, feedforward DRC 404 is slowed directly throughadjusting DRC 404's attack and/or release time constants. This slowedcompression, in both cases, results in the reduction of the spectralspread of harmonic distortion and intermodulation distortion productsrelative to the prior art.

A time constant τ for direct slowing of the feedforward DRC 404 may bedetermined based on a range of frequencies that are subjected to thesignal processing, e.g., the range of frequency bands output by thespectral decomposition. In some embodiments, this range may extend from60 Hz to 20 kHz. The time constant τ for a corresponding frequency f isgiven by τ=1/(2πf). Thereby, the range of the frequency bands can betranslated into a range for the time constant τ for direct slowing ofthe feedforward DRC 404. For the example of a range of frequenciesextending from 60 Hz zo 20 kHz, a corresponding range for the timeconstant τ would extend from about 0.008 ms to about 2.65 ms. Thus, insome embodiments the range for the time constant τ may be chosen toextend from 0.01 ms to 3 ms. The actual time constant τ that is thenused for direct slowing of the feedforward DRC 404 may be chosen fromthe range(s) for the time constant. The attack time constant and/or therelease time constant for directly slowing the feedforward DRC 404 maybe calculated from the selected time constant τ, e.g., in the mannerdescribed in Fred Floru, Attack and Release Time Constants in RMS-BasedCompressors and Limiters, 99^(th) AES Convention, 6-9 Oct. 1999.

Notably, the above range for the time constant τ is compatible withtypical update rates of the parameters of the feedforward DRC 404. Forexample, updating the feedforward DRC 404 every 64 samples for samplingrates of 44,100 Hz and 48,000 Hz will yield update intervals of 1.45 msand 1.33 ms, respectively, which fall into the above range for the timeconstant τ.

In some embodiments, the time constant τ for slowing of the feedforwardDRC 404 in a given frequency band may depend on a frequency f within therespective frequency band. For example, the frequency f may be chosen tobe a characteristic frequency f_(c) of the RC filter (e.g., high-pass,low-pass, or bandpass filter) for the respective frequency band. Thischaracteristic frequency f_(c) may be, for example, the lower cutofffrequency, the upper cutoff frequency, or the center frequency of the RCfilter (or likewise, the lower cutoff frequency, the upper cutofffrequency, or the center frequency of the respective frequency band).Again, the time constant τ for the given frequency band can bedetermined via τ=1/(2πf) or τ=1/(2πf_(c)) and the attack time constantand/or the release time constant for directly slowing the feedforwardDRC 404 may be calculated from the time constant τ, e.g., in the mannerdescribed in Fred Floru, Attack and Release Time Constants in RMS-BasedCompressors and Limiters, 99^(th) AES Convention, 6-9 Oct. 1999. In someembodiments, the time constant may be the RC time constant of the RCfilter, τ=RC.

As noted above, indirect slowing of the feedforward DRC 404 may involvesignal oversampling by a factor N (e.g., 128, 256, etc.). For example,the signal processing may be performed in the FFT domain, after applyingan n-point FFT (e.g., 256 point FFT or 512 point FFT). The n-point FFTmay be applied in each frequency band, for example, or the spectraldecomposition may operate in the FFT domain. The transforms in eachfrequency band may then be overlapped by n/N samples (e.g., by 2 samplesfor a 256 point FFT and an oversampling rate of N=128). This means thata rate N times higher than the theoretical (sub-band) sample rate isused. However, this rate is still by a rate of n/N slower than the fulldata rate in the respective frequency band. This relative slowness meansthat the feedforward DRC 404 behaves like a full-rate instantaneous DRCwith attack and release time constants applied. Incidentally, applyingthe n-point FFT drastically reduces the sample rate (e.g., in eachfrequency band) and thereby also allows for significant processingsavings over a time domain implementation. For a given n, theoversampling rate N may range from 1 to n/2, for example (whichtranslates into an overlap in the range between n/2 samples and 2samples). For typical implementations (e.g., n=256, 512, 1024), theoversampling rate N may be in the range from 128 to 512, for example.

As noted above, slowing the first DRC relative to an instantaneous DRCmay relate to both oversampling the respective subband signal as well asto increasing attack and/or release time constants of the first DRC(i.e., setting the attack and/or release time constants to valuesdifferent from 0). One of ordinary skill in the art may appreciate thatslowing the first DRC may be achieved through the combination of bothindirect slowing (i.e. oversampling) and direct slowing (i.e., alteringthe time constants of the first DRC). For direct slowing, τ is relatedto the cutoff frequency f_(c), an alternative parameter of the RCcircuit, by τ=RC=1/(2π f_(c)). An (indirect) equivalent of the timeconstant τ of the directly slowed first DRC for the slowing byoversampling can be calculated by dividing the oversampling rate N bythe sampling rate (e.g. 44100 Hz). To this extent, the combinatorialeffect of indirect and direct slowing of the DRC is readily calculableas a function of these two values.

Importantly, slowing the feedforward DRC 404 also replaces the need forhaving output band pass filter 309—which results in a more symmetricaldistribution of the harmonic distortion that remains. In the healthyhearing system, natural distortion emanates symmetrically at allfrequency regions of the basilar membrane as the cochlear process, initself, can be thought of as a resonant system with a continuum ofchanging properties. As seen in FIG. 8, when the distortion pattern of agiven frequency is moved towards one of the band edges of an output bandpass filter, see panels 802, 803, the distortion pattern isasymmetrically curtailed, see panel 802. When an output band pass filteris not present, harmonic symmetry is better retained—thus betteremulating the natural distortion creating by the basilar membrane.

From the compression output 405, the audio signal is processed by afeedback DRC 406. The feedback DRC 406 is delayed relative to thefeedforward DRC. The feedback pathway is tapped from the output of thefeedforward DRC process. The feedback pathway may be attenuated bythresholding to obtain signal parts above a certain threshold. Thissignal may then be low-pass filtered for temporal smoothing and may bemultiplied by a scalar factor. The aforementioned delay may be achievedthrough the use of a buffer, such as a ring buffer, for example. Thisresults in a stream of attenuation values that by their delays simulatethe synaptic delays of the MOC feedback system. This stream of values issubsequently used to modulate the audio signal provided to thefeedforward DRC 404 within each band. Modulation, feedforwardcompression and feedback compression proceed in a continuous manner.Thus, the feedback loop dynamically adapts compression to the audiosignal level, enabling more effective mitigation of off-frequency soundmasking—a process that physiologically occurs in the auditory system.

Frequency bands may be modulated by a gain 408 and, finally, recombinedin operator 503 to form a full wide audio band signal again to beprovided at the control output 504. Each frequency band may have itsown, distinct parameters, e.g. gain, attenuation factors, etc.

In some embodiments, as shown in FIG. 4B, a dynamic threshold can becreated in feedforward DRC 404 by directly acting on feedforward DRC 404with a controller 416, which may involve adapting, under control of thecontroller 416, the compression function that is applied by thefeedforward DRC. Adapting the compression function can comprise adaptingthe parameters of the compression function. This is in contrast to theexample of FIG. 4A, which utilizes feedback compressor 406 and upstreammodulator 407, acting as an attenuator, to achieve various dynamicthresholds in feedforward DRC 404 as described above. As such, providingfor a dynamic threshold of the feedforward DRC 404 could be said to beequivalent to modulating (e.g., attenuating) the subband signal upstreamof the feedforward DRC 404. With respect to FIG. 4B, controller 416 canbe configured in hardware circuitry, in software, or some combination ofthe two, such that controller 416 generates and applies one or morecontrol signals to cause internal adjustments to feedforward DRC 404,where the internal adjustments correspond to setting the dynamicthreshold value of feedforward DRC 404 to the desired configuration.These control signal can be static in nature (e.g. applied at a singlepoint in time, applied at fixed points in time or in fixed intervals,etc.) or can be dynamic in nature (e.g. applied continuously over a timeinterval, updated or modified based on the measured dynamic thresholdactually achieved in feedforward DRC 404 at prior points in the timeinterval [e.g. T-1, T-2, . . . , T-n], etc.) In some embodiments, thedynamic threshold that is commanded by controller 416 can bepre-configured, for example stored in a memory or instruction storeassociated with or otherwise communicatively coupled to controller 416and/or feedforward DRC 404. The commanded dynamic threshold mayadditionally or alternatively be generated based on one or more userinputs received at controller 416. In addition to controlling thedynamic threshold of feedforward DRC 404, controller 416 canadditionally or alternatively be employed to provide dynamic controlover one or more of a feedforward ratio, an attack time constant, arelease time constant, and/or a soft knee function of the feedforwardDRC 404.

FIG. 4C depicts a graph of different example dynamic thresholds that maybe created or applied to feedforward DRC 404 (i.e., respectivecompression functions may be applied to the subband signal by thefeedforward DRC 404). Notably, the various different dynamic thresholdsof FIG. 4C may be created or applied to feedforward DRC 404 by eitherone of the system of FIG. 4A and the system of FIG. 4B, recalling thatthe two systems can provide identical or substantially identical dynamicthreshold control despite their differences in design.

As mentioned above, direct and/or indirect slowing can be applied tofeedforward DRC 404 in order to control or minimize distortion of theaudio signal and audio signal processing pathway. In some embodiments, asoft knee function can be added to the feedforward compressor (e.g.feedforward DRC 404) in order to achieve a same or similar effect inlieu of using direct or indirect slowing. A soft knee function may alsobe added to the feedforward compressor/feedforward DRC 404 in order toaugment the distortion minimization effect that is already achieved bythe use of direct and/or indirect slowing as described previously.

FIG. 4D illustrates an example soft knee function that can be applied tofeedforward DRC 404 (i.e., to the compression function of thefeedforward DRC 404). The soft knee function can be provided as alogarithmic function or other forms as appreciated by one of ordinaryskill in the art. For example, a soft knee point (and corresponding softknee region, see FIG. 4D) might be present in a logarithmic soft 10function that takes the form f(x)=a*In(b*x)+C, where x is the inputsignal and the values of a, b and C are determined by the ratio, thethreshold, and how hard the knee is. For example, assuming the thresholdto be denoted as T, the ratio as R, and a soft knee width spanning x₁(with x₁<T), to x₂ (with x₂>T), then the compression function F(x) maybe given by F(x)=x for 0≤x<x₁, by F(x)=a*In(b*x)+C for x₁≤x<x₂, and byF(x)=R*(x−x₂)+R*(x₂−T)+T for x₂≤x, where a, b, and C are determined suchthat function F(x) is continuous at x₁ and x₂, with F(x₁)=x₁ andF(x₂)=T+R*(x₂−T). Here, W=x₂−x₁ is the knee width. Preferably, the softknee is symmetric with respect to the threshold T, i.e., x₁=T−W/2 andx₂=T+W/2. As seen in the graph of FIG. 4D, the application of the softknee function at feedforward DRC 404 smooths the oblique junction pointor corner that is otherwise observed in the non-linear broken stickcompression configuration (e.g. the “hard knee”). By eliminating thissharp intersection, the soft knee function eliminates the hard linearbreak that otherwise is present. This linear break has a perceptuallyunpleasant distortive effect on the signal that otherwise is present andperceived as distortion. Accordingly, application of the soft kneefunction at feedforward DRC 404 can minimize unpleasant distortion,either functioning alone or in combination with the distortion reductionachieved by applying direct and/or indirect slowing to feedforward DRC404.

Similar to how the use of direct and/or indirect slowing replaces theneed for an output band pass filter (such as output band pass filter309, used in conventional systems), the use of a soft knee function canlikewise eliminate the need for an output band pass filter.Advantageously, the elimination of the output band pass filter resultsin a more symmetrical distribution of the harmonic distortion thatremains. Importantly, slowing the feedforward DRC 404 also replaces theneed for having output band pass filter 309—which results in a moresymmetrical distribution of the harmonic distortion that remains. In thehealthy hearing system, natural distortion emanates symmetrically at allfrequency regions of the basilar membrane as the cochlear process, initself, can be thought of as a resonant system with a continuum ofchanging properties. As seen in FIG. 8, when the distortion pattern of agiven frequency is moved towards one of the band edges of an output bandpass filter (e.g., see 802, 803), the distortion pattern isasymmetrically curtailed, see panel 802. When an output band pass filteris not present, harmonic symmetry is better retained—thus betteremulating the natural distortion created by the basilar membrane.

As referenced above, FIGS. 5A and 5B illustrate example processes ofspectral decomposition 501. In FIG. 5A, frequency bands may be modulatedby a gain (e.g. gain 408) and/or attenuation factor, and, finally,recombined in operator 503 to form a full wide audio band signal againto be provided at the control output 504. Each frequency band may haveits own, distinct parameters, e.g. gain, attenuation factors, etc.

In some embodiments, an asymmetric soft clipping function 506 (i.e. anasymmetric clipping function with a soft clipping component) can beadded to the control output 504, as depicted in FIG. 5B. More generally,it is appreciated that asymmetric soft clipping function 506 can beadded at the output of one or more multiband compressor systems of thepresent disclosure. The use of soft clipping in an asymmetric softclipping function 506 minimizes or eliminates the harsh sounding effectsof over-saturation of one or more amplifiers. The soft clippingcomponent of the asymmetric soft clipping function 506 does so bysmoothing hard edges that are otherwise created when the over-saturatedamplification causes clipping in the output signal. As compared to asymmetric clipping function, which saturates with similar thresholds onthe positive and negative parts of the waveform, an asymmetric clippingfunction (such as asymmetric soft clipping function 506) saturates withdifferent thresholds on the positive and negative parts of the waveform.As a result of this asymmetry, both even and odd harmonics in thecontrol output 504 are emphasized, and audio signals in which even andodd harmonics are emphasized are perceived as pleasant to listen to. Bycomparison, a symmetric clipping function would emphasize only oddharmonics, which are known to be perceived as harsh sounding andunpleasant to listen to. In some embodiments, asymmetric soft clippingfunction 506 can be provided by one or more cubic functions, arctangentfunctions, or some combination of the two.

Another example embodiment of the invention is illustrated in FIG. 9, inwhich sound processed according to one or more of the configurationsdescribed above with respect to FIGS. 4A-D and FIGS. 5A-B is split intoa first and second signaling pathway (or signal pathway). Specifically,a wide band audio signal is provided at processing input 901 and thendivided into a first pathway (first signal pathway) 902 and a secondpathway (second signal pathway) 903. In this example, the second pathway903 is only subject to a delay 904 and a protective limiter 905. Incontrast, in the first pathway 902, the audio signal from the controlinput 901 is spectrally decomposed and processed according to theconfiguration of FIG. 4. Each pathway 902, 903 may include a weightingoperator 906 and 907, respectively. For example, these weightingoperators 906 and 907 may be correlated by a common function that may beadjustable by a user by one single control variable 910. Then thesepathways 902 and 903 are recombined according to their weighting factorsin operator 908 and provided to the processing output 909.

Parallel compression provides the benefit of allowing the user to mix‘dry’ unprocessed or slightly processed sound with ‘wet’ processedsound, enabling customization of processing based on subjectivepreference. For example, this enables hearing impaired users to use ahigh ratio of heavily processed sound relative to users with moderate tolow hearing loss. Furthermore, by reducing the dynamic range of an audiosignal by bringing up the softest sounds, rather than reducing thehighest peaks, it provides audible detail to sound. The human ear issensitive to loud sounds being suddenly reduced in volume, but lesssensitive to soft sounds being increased in volume, and this mixingmethod takes advantage of this observation, resulting in a more naturalsounding reduction in dynamic range compared with using a dynamic rangecompressor in isolation. Additionally, parallel compression is inparticular useful for speech-comprehension and/or for listening to musicwith full, original timbre.

To mix two different signal pathways requires that the signals in thepathways conform to phase linearity, or into the pathway's identicalphase using phase distortion, or the pathway mixing modulator involves aphase correction network in order to prevent any phase cancellationsupon summing the correlated signals to provide an audio signal to thecontrol output. Notably, parallel compression is problematic using theapproach in the prior art as the recursive input and output IIR bandpass filters introduce phase distortion into the audio signal.Superposition of a phase-distorted signal with the correlated, originalaudio signal can cause so-called comb-filtering effects, which adverselyaffects the timbral quality of the results. Users are sensitive to theseeffects, which are detrimental to the subjective hearing experience.

A further example embodiment of the invention is illustrated in FIG. 10,in which a wide band audio signal is provided at processing input 1001and then divided into a first processed pathway 1002 (first signalingpathway, or first signal pathway), utilizing the same processingconfiguration as FIG. 3, and a second processed pathway 1003 (firstsignaling pathway, or first signal pathway). Similar to theconfiguration in FIG. 9, the phases of each processed pathway must be insync in order to prevent phase cancellation and comb-filtering effects.The second processed pathway 1003 may include, for instance, a soundenhancing algorithm, such as a speech-comprehension algorithm, to allowthe user to adjust pathway ratios to allow for better speechcomprehension and/or to have a subjectively more comfortable musichearing experience for the user in a respective background noiseenvironment. A further example could also include a differentlyparameterized process using the configuration illustrated in FIG. 4Aand/or FIG. 4B. Generally, any sound processing algorithm, such as ahearing aid algorithm, preferably having linear phase characteristics,could be mixed with the first processed pathway 1002. The two processedpathways have to be in line with one of the following rules. The signalsin each pathway must either (i) conform to phase linearity, (ii)introduce identical phase distortions, or (iii) involve a phasecorrection network in order to prevent any phase cancellations uponsumming the correlated signals to form the output. Preferably, option(i) or option (ii) is implemented as this allows for lean and simpleimplementation.

A further example embodiment of the invention is illustrated in FIGS. 11and 12, in which a wide band audio signal is provided at processinginput 1101, 1201 and then spectrally decomposed into a plurality offrequency bands (e.g., into a plurality of subband signals, each subbandsignal in a respective frequency band). Spectral decomposition may beperformed by the input band pass filter 1202, for example. Eachrespective frequency band (e.g., subband signal) is divided into a firstpathway (first signal pathway) 1103 and a second pathway (second signalpathway) 1104. In this example, the second pathway 1104 is lightlyprocessed as it only includes a delay 1105 and a protective limiter1106. In contrast, first pathway 1103 is processed similarly to theconfiguration illustrated in FIGS. 5A-B. Namely, the first pathwayfrequency band is provided at a compression input, which is feed forwardcompressed by feedforward DRC 404. From the compression output, theaudio signal is processed by a feedback DRC 406, wherein the feedbackDRC 406 may be delayed relative to the feedforward DRC 404. That is, theoutput of the feedback processing pathway may be deliberately delayed,e.g. by a delay element (such as a buffer, for example). The delay maybe inserted before or after the feedback DRC. Subsequently, thecompressed audio signal may be modulated via a modulator 407 toattenuate the audio signal provided to feedforward DRC 404. Thecompressed frequency bands may then be modulated by a gain 408, 1107.First and second frequency band pathways may include a weightingoperator 1108 and 1109, respectively. Here, these weighting operators1108 and 1109 may be correlated by a common function that may beadjustable by a user by one single control variable 1112. Then thesepathways 1103 and 1104 are recombined according to their weightingfactors in operator 1110 and provided to the processing output 1111.Finally, the frequency bands (e.g., subband signals) are recombined inoperator 1203 to form a full wide audio band signal again to be providedat the control output 1204. The configuration of FIGS. 11 and 12importantly allows the user much more control over which frequencies areprocessed in the audio spectrum of a signal. For instance, in a musiccomposition with intense treble, it may be preferable to process soundsacross a higher frequency range. Conversely, while focusing on humanspeech, approximately around 150 Hz-4 kHz, processing may focus onnarrower spectra. Generally, spectral processing can be adjusted forcomputational savings purposes.

A further example embodiment of the invention is illustrated in FIG. 13.A wide band audio signal is provided at processing input 1301 and thenspectrally decomposed into a plurality of frequency bands (e.g., into aplurality of subband signals, each subband signal in a respectivefrequency band). Spectral decomposition may be performed by the inputband pass filter 1302, for example. Each respective frequency band isprovided at a compression input 1303, which is feed forward compressedby feedforward DRC 1304. Feedforward DRC 1304 is slowed relative to aninstantaneous DRC. This may occur directly or indirectly, e.g., in themanner described above. From the compression outputs 1305 of a pluralityof frequency bands 1306, wherein each compression output from arespective frequency band is assigned an individual weighting 1307,1308, 1309, the respective audio signals are processed by feedbackcompression 1314. Although FIG. 13 shows the feeding of weightedfeedforward compressed audio signals from neighboring frequency bandsfor a single frequency band only, it is understood that feedbackcompression in each frequency band may receive weighted feedforwardcompressed audio signals as inputs from respective neighboring frequencybands. The feedback DRC 1314 may be delayed relative to the feedforwardDRC 1304. That is, the output of the feedback DRC may be deliberatelydelayed, e.g., by a delay element (such as a buffer for example). Thedelay may be inserted before or after the feedback DRC. The plurality ofdelayed feed-back compressed audio signals may be modulated via amodulator 1310 to attenuate the audio signal provided to feedforward DRC1304. The compressed frequency bands may then be modulated by a gain1311 and, finally, recombined in operator 1312 to form a full wide audioband signal again to be provided at the control output 1313. The abilityto attenuate the audio signal in a given frequency band as a functionthat includes signal levels in one or more neighboring frequency bands,provides a more refined degree of parameterization for an augmentedhearing experience. Moreover, this process happens naturally in theolivocochlear system in the ear (Effects of electrical stimulation ofefferent olivochoclear neurons on cat auditory-nerve fibers. Ill, Tuningcurves and thresholds at CF, Guinan & Gifford, 1988)—and thus this audioprocessing configuration more closely models healthy hearing in theauditory stem.

FIG. 14 shows an example of computing system 1400 (e.g., audio device,smart phone, etc.) in which the components of the system are incommunication with each other using connection 1405. Connection 1405 canbe a physical connection via a bus, or a direct connection intoprocessor 1410, such as in a chipset architecture. Connection 1405 canalso be a virtual connection, networked connection, or logicalconnection.

In some embodiments computing system 1400 is a distributed system inwhich the functions described in this disclosure can be distributedwithin a datacenter, multiple datacenters, a peer network, etc. In someembodiments, one or more of the described system components representsmany such components each performing some or all of the function forwhich the component is described. In some embodiments, the componentscan be physical or virtual devices.

Example system 1400 includes at least one processing unit (CPU orprocessor) 1410 and connection 1405 that couples various systemcomponents including system memory 1415, such as read only memory (ROM)and random access memory (RAM) to processor 1410. Computing system 1400can include a cache of high-speed memory connected directly with, inclose proximity to, or integrated as part of processor 1410.

Processor 1410 can include any general-purpose processor and a hardwareservice or software service, such as services 1432, 1434, and 1436stored in storage device 1430, configured to control processor 1410 aswell as a special-purpose processor where software instructions areincorporated into the actual processor design. Processor 1410 mayessentially be a completely self-contained computing system, containingmultiple cores or processors, a bus, memory controller, cache, etc. Amulti-core processor may be symmetric or asymmetric.

To enable user interaction, computing system 1400 includes an inputdevice 1445, which can represent any number of input mechanisms, such asa microphone for speech, a touch-sensitive screen for gesture orgraphical input, keyboard, mouse, motion input, speech, etc. In someexamples, the input device can also include audio signals, such asthrough an audio jack or the like. Computing system 1400 can alsoinclude output device 1435, which can be one or more of a number ofoutput mechanisms known to those of skill in the art. In some instances,multimodal systems can enable a user to provide multiple types ofinput/output to communicate with computing system 1400. Computing system1400 can include communications interface 1440, which can generallygovern and manage the user input and system output. In some examples,communication interface 1440 can be configured to receive one or moreaudio signals via one or more networks (e.g., Bluetooth, Internet,etc.). There is no restriction on operating on any particular hardwarearrangement and therefore the basic features here may easily besubstituted for improved hardware or firmware arrangements as they aredeveloped.

Storage device 1430 can be a non-volatile memory device and can be ahard disk or other types of computer readable media which can store datathat are accessible by a computer, such as magnetic cassettes, flashmemory cards, solid state memory devices, digital versatile disks,cartridges, random access memories (RAMs), read only memory (ROM),and/or some combination of these devices.

The storage device 1430 can include software services, servers,services, etc., that when the code that defines such software isexecuted by the processor 1410, it causes the system to perform afunction. In some embodiments, a hardware service that performs aparticular function can include the software component stored in acomputer-readable medium in connection with the necessary hardwarecomponents, such as processor 1410, connection 1405, output device 1435,etc., to carry out the function.

For clarity of explanation, in some instances the present technology maybe presented as including individual functional blocks includingfunctional blocks comprising devices, device components, steps orroutines in a method embodied in software, or combinations of hardwareand software.

The presented technology creates improved, biologically-inspired DSPalgorithms that more closely mimic the functional processing of thehealthy human ear. The invention avoids the limitations inherent inprior art DSP methodologies, namely poorly constrained frequencydistortion and phase distortion. To this extent, the invention providesan enhanced listening experience both to hard of hearing individuals aswell as individuals with healthy hearing, who experience a richer,crisper listening experience of audio content.

Further example embodiments of the disclosure are summarized in theEnumerated Example Embodiments (EEEs) listed below.

A first EEE relates to a method for processing an audio signal forreplay on an audio device, the method comprising: a) performing aspectral decomposition of the audio signal (501) into a plurality ofsubband signals using a band pass filter (402, 502); b) for each subbandsignal, providing the subband signal to a respective modulator (407) andfrom the modulator output, providing the subband signal to a respectivefirst processing path that includes a first dynamic range compressor,DRC (404); c) for each subband signal, feedforward compressing thesubband signal by the respective first DRC (404) to obtain afeedforward-compressed subband signal; d) for each subband signal,providing the feedforward-compressed subband signal to a secondprocessing path that includes a second DRC (406), compressing thefeedforward-compressed subband signal by the respective second DRC(406), and providing an output of the second processing path to therespective modulator (407), wherein modulating the subband signal by therespective modulator (407) is performed in dependence on the output ofthe second processing path; and e) recombining thefeedforward-compressed subband signals, wherein feedforward compressingcomprises, for each subband signal, slowing the respective first DRC(404) relative to an instantaneous DRC.

A second EEE relates to a method of processing an audio signal forreplay on an audio device, the method comprising dividing an unprocessedaudio signal into a first signal pathway (903, 1003) and a second signalpathway (902, 1002), processing the audio signal in the first signalpathway (902, 903), and recombining outputs of the first and secondsignal pathways (902/903, 1002/1003) at a ratio (910, 1004), wherein theprocessing in the first signal pathway (902, 1002) comprises: a)performing a spectral decomposition of the audio signal (501) into aplurality of subband signals using a band pass filter (402, 502); b) foreach subband signal, providing the subband signal to a respectivemodulator (407) and from the modulator output, providing the subbandsignal to a respective first processing path that includes a firstdynamic range compressor, DRC (404); c) for each subband signal,feedforward compressing the subband signal by the respective first DRC(404) to obtain a feedforward-compressed subband signal; d) for eachsubband signal, providing the feedforward-compressed subband signal to asecond processing path that includes a second DRC (406), compressing thefeedforward-compressed subband signal by the respective second DRC(406), and providing an output of the second processing path to therespective modulator (407), wherein modulating the subband signal by therespective modulator (407) is performed in dependence on the output ofthe second processing path; and e) recombining thefeedforward-compressed subband signals, wherein feedforward compressingcomprises, for each subband signal, slowing the respective first DRC(404) relative to an instantaneous DRC.

A third EEE relates to the method of the second EEE, wherein the ratio(910, 1004) is a user-defined ratio.

A fourth EEE relates to the method according to the second or thirdEEEs, wherein the second signal pathway (903) features a delay and thedelayed signal is subjected to a protective limiter.

A fifth EEE relates to the method according to EEEs 2-4, whereinfrequencies only between 125 Hz and 12,000 Hz are processed in the firstsignal pathway (902, 1002).

A sixth EEE relates to a method for processing an audio signal forreplay on an audio device, the method comprising: a) performing aspectral decomposition (1202) of the audio signal into a plurality ofsubband signals using a band pass filter (1102, 1202); b) for eachsubband signal, dividing the subband signal into a first signal pathway(1103) and a second signal pathway (1103), processing the subband signalin the first signal pathway (1103), and recombining the first and secondsignal pathways (1103, 1104) at a ratio to obtain a processed subbandsignal; and c) recombining the processed subband signals, wherein, foreach subband signal, the processing of the subband signal in the firstsignal pathway (1103) comprises: b1) providing the subband signal to arespective modulator (407) and from the modulator output, providing thesubband signal to a respective first processing path that includes afirst dynamic range compressor, DRC (404); b2) feedforward compressingthe subband signal by the respective first DRC (404) to obtain afeedforward-compressed subband signal; and b3) providing thefeedforward-compressed subband signal to a second processing path thatincludes a second DRC (406), compressing the feedforward-compressedsubband signal by the respective second DRC (406), and providing anoutput of the second processing path to the respective modulator (407),wherein modulating the subband signal is performed in dependence on theoutput of the second processing path, and wherein feedforwardcompressing comprises, for each subband signal, slowing the respectivefirst DRC (404) relative to an instantaneous DRC.

A seventh EEE relates to the method accord to the sixth EEE, wherein thesecond signal pathway (1104) features a delay and the delayed signal issubjected to a protective limiter.

An eighth EEE relates to a method for processing an audio signal forreplay on an audio device, the method comprising: a) performing aspectral decomposition (1302) of the audio signal into a plurality ofsubband signals using a band pass filter; b) for each subband signal,providing the subband signal to a respective modulator (407) and fromthe modulator output, providing the subband signal to a respective firstprocessing path that includes a first dynamic range compressor, DRC(1304); c) for each subband signal, feedforward compressing the subbandsignal by the respective first DRC (1304) to obtain afeedforward-compressed subband signal; d) for each subband signal,providing the feedforward-compressed subband signal to a secondprocessing path that includes a second DRC (1314) and further providingone or more feedforward-compressed subband signals from neighboringfrequency bands, each weighted with a respective weighting factor, tothe second processing path, compressing, in the second processing path,the feedforward-compressed subband signal and the weightedfeedforward-compressed subband signals from the neighboring frequencysubbands by the respective second DRC (1314), and providing an output ofthe second processing path to the respective modulator, whereinmodulating the subband signal is performed in dependence on the outputof the second processing path; and e) recombining thefeedforward-compressed audio signals, wherein feedforward compressingcomprises, for each subband signal, slowing the respective first DRC(1304) relative to an instantaneous DRC.

A ninth EEE relates to a method according to any of the preceding EEE's,further comprising, for each subband signal, delaying the output of therespective second processing path (406).

A tenth EEE relates to a method according to any of the preceding EEE's,wherein, for each subband signal, the output of the respective secondprocessing path is delayed by a delay amount that is in the intervalfrom 5 ms to 20 ms.

An eleventh EEE relates to a method according to any of the precedingEEE's, wherein the band pass filter (402, 502) is phase linear.

A twelfth EEE relates to a method according to any of the precedingEEE's, wherein the band pass filter (402, 502) is a finite impulseresponse filter operating in the frequency domain.

A thirteenth EEE relates to a method according to any of the precedingEEE's, wherein the first DRC (404) is slowed by multi-rate signalprocessing as part of the spectral decomposition process.

A fourteenth EEE relates to a method according to any of EEE's 1-12,wherein the first DRC (404) is slowed by changing the attack and/orrelease time constants of the first DRC (404).

A fifteenth EEE relates to a method according to any of the precedingEEE's, wherein the audio device is one of: a mobile phone, a tablet, acomputer, a television, a pair of headphones, a hearing aid or a speakersystem.

A sixteenth EEE relates to a method according to any of the precedingEEE's, wherein only one bandpass filter is employed per frequency band.

A seventeenth EEE relates to an audio output device comprising: aprocessor for processing an audio signal according to the methods of anyof the preceding EEE's.

An eighteenth EEE relates to a computer readable storage medium storinga program causing an audio output device to execute audio processingaccording to the methods of any of EEE's 1 to 16.

LIST OF REFERENCE NUMERALS

201 outer- and middle ear

202 filterbank

203 inner hair cell (IHC)

204 IHC synapse

205 auditory nerve

206 acoustic reflex

207 MOC reflex

208 brainstem

301 control input

302 spectral decomposition

303 IIR input band pass filter

304 compression input

305 instantaneous DRC

306 compression output

307 feedback DRC

308 modulator

309 IIR output band pass filter

310 gain

311 operator

401 control input

402 input bandpass filter

403 compression input

404 feedforward DRC

405 compression output

406 feedback DRC

407 modulator

408 gain

501 spectral decomposition

502 input bandpass filter

503 operator

504 control output

601 spectrogram-distortion from instantaneous compression (IC) output (1kHz input)

602 spectrogram-distortion from IC output with input and output IIR bandpass filters

603 spectrogram-distortion from DRC with input FIR band pass filter

701 Psychophysical tuning curve from a subject before and after exposureto algorithm according to claim 1—left ear

702 Psychophysical tuning curve from a subject before and after exposureto algorithm according to claim 1—right ear

801 Distortion going through a band pass filter—centered

802 Distortion going through a band pass filter—shifted from center

803 Distortion—no band pass filter—centered

804 Distortion—no band pass filter—shifted from center

901 Control input

902 Processed pathway

903 Unprocessed pathway

904 delay

905 protective limiter

906 processed pathway weighting operator

907 unprocessed pathway weighting operator

908 recombining the two pathways

909 control output

910 single control variable

1001 control input

1002 processed pathway 1

1003 processed pathway 2

1004 control variable

1101 control input

1102 input bandpass filter

1103 processed frequency band pathway

1104 unprocessed frequency band pathway

1105 delay

1106 protective limiter

1107 gain

1108 processed frequency band pathway weighting operator

1109 unprocessed frequency band pathway weighting operator

1110 single control variable

1111 control output

1201 control input

1202 input band pass filter

1203 recombining frequency bands

1204 control output

1301 control input

1302 input bandpass filter

1303 compression input of primary frequency band

1304 feedforward DRC

1305 feedback DRC

1306 alternate frequency bands

1307 weighting 1

1308 weighting 2

1309 weighting 3

1310 modulator

1311 gain

1312 recombining the frequency bands

1313 control output

1314 feedback DRC

What is claimed is:
 1. A method of processing an audio signal for replayon an audio device, the method comprising: performing a spectraldecomposition of the audio signal into a plurality of subband signalsusing band pass filters; for each subband signal of the plurality ofsubband signals: providing the subband signal to a first processing paththat includes a first dynamic range compressor (DRC); adjusting, via acontroller, a dynamic threshold associated with the first DRC, whereinthe controller generates and applies one or more control signals to thefirst DRC to adjust the dynamic threshold; and feedforward compressingthe received subband signal by the first DRC to obtain afeedforward-compressed subband signal, wherein thefeedforward-compressed subband signal is obtained based at least in parton the adjusted dynamic threshold of the first DRC; and recombining thefeedforward-compressed subband signals.
 2. The method of claim 1,wherein adjusting the dynamic threshold associated with the first DRCcomprises directly modifying the first DRC in order to obtain a desiredadjustment to the dynamic threshold associated with the first DRC. 3.The method of claim 2, wherein the first DRC is directly modified by acontroller communicatively coupled to the first DRC.
 4. The method ofclaim 1, further comprising performing distortion control at the firstDRC, wherein the distortion control is performed without the use of anoutput band pass filter.
 5. The method of claim 4, wherein performingdistortion control at the first DRC comprises applying a soft kneefunction to the first DRC, wherein applying the soft knee functionsmooths a non-linear broken stick compression configuration.
 6. Themethod of claim 5, wherein one or more parameters of the soft kneefunction are adjusted based on one or more of the dynamic thresholdassociated with the first DRC and a measurement of the non-linear brokenstick compression configuration.
 7. The method of claim 5, whereinperforming distortion control further comprises slowing the first DRCrelative to an instantaneous DRC by oversampling the subband signal orby increasing attack and/or release time constants of the first DRCrelative to an instantaneous DRC.
 8. The method of claim 1, furthercomprising providing the recombined feedforward-compressed subbandsignals to a soft clipping function located downstream from the firstprocessing path, wherein the soft clipping function emphasizes even andodd order harmonics in the recombined feedforward-compressed subbandsignals.
 9. The method of claim 8, wherein the soft clipping function isan asymmetric soft clipping function.
 10. The method of claim 1, whereinadjusting the dynamic threshold associated with the first DRC comprisesslowing the feedforward compression performed by the first DRC, relativeto an instantaneous DRC.
 11. The method of claim 10, further comprising,for each subband signal of the plurality of subband signals: providingthe subband signal to a modulator upstream from the first DRC; from themodulator output, providing the modulated subband signal to the firstprocessing path as the received subband signal for feedforwardcompressing by the first DRC; providing the feedforward-compressedsubband signal to a second processing path that includes a second DRC;compressing the feedforward-compressed subband signal by the second DRC;and providing an output of the second processing path to the modulator,wherein modulating the subband signal by the modulator is performed independence on the output of the second processing path.
 12. The methodof claim 11, further comprising providing the recombinedfeedforward-compressed subband signals to a soft clipping functionlocated downstream from the first processing path, wherein the softclipping function emphasizes even and odd order harmonics in therecombined feedforward-compressed subband signals.
 13. The method ofclaim 12, wherein the soft clipping function is an asymmetric softclipping function.
 14. The method of claim 11, further comprising:dividing an unprocessed audio signal into a first signal pathway and asecond signal pathway; processing the unprocessed audio signal in thefirst signal pathway; and recombining outputs of the first signal andsecond signal pathways at a ratio.
 15. The method of claim 14, wherein:the ratio is a user-defined ratio; the second signal pathway features adelay and the delayed signal is subjected to a protective limiter; andfrequencies between 60 Hz and 20,000 Hz are processed in the firstsignal pathway.
 16. The method of claim 11, further comprising, for eachsubband signal of the plurality of subband signals: providing one ormore feedforward-compressed subband signals from neighboring frequencybands, each weighted with a weighting factor, to the second processingpath; and compressing, in the second processing path, a signal obtainedby adding the feedforward-compressed subband signal and the weightedfeedforward-compressed subband signals from the neighboring frequencysubbands, by the second DRC.
 17. An audio output device comprising: atleast one processor; and at least one memory storing instructions, whichwhen executed causes the at least one processor to: perform a spectraldecomposition of an audio signal into a plurality of subband signalsusing a band pass filter; for each subband signal of the plurality ofsubband signals: provide the subband signal to a first processing paththat includes a first dynamic range compressor (DRC); adjust, via acontroller, a dynamic threshold associated with the first DRC, whereinthe controller generates and applies one or more control signals to thefirst DRC to adjust the dynamic threshold; and feedforward compress thereceived subband signal by the first DRC to obtain afeedforward-compressed subband signal based at least in part on theadjusted dynamic threshold of the first DRC; and recombine thefeedforward-compressed subband signals.
 18. The audio output device ofclaim 17, wherein adjusting the dynamic threshold associated with thefirst DRC comprises directly modifying the first DRC in order to obtaina desired adjustment to the dynamic threshold associated with the firstDRC.
 19. The audio output device of claim 18, wherein the instructionsfurther cause the at least one processor to perform distortion controlat the first DRC by applying a soft knee function to the first DRC,where at least one parameter of the soft knee function is adjusted basedat least in part on the dynamic threshold associated with the first DRC.20. The audio output device of claim 17, wherein the instructionsfurther cause the at least one processor to provide the recombinedfeedforward-compressed subband signals to a soft clipping functionlocated downstream from the first processing path, wherein the softclipping function emphasizes even and odd order harmonics in therecombined feedforward-compressed subband signals.